1. Field of the Invention
This invention relates to a method, an apparatus, and a computer program for processing a digital audio signal to improve a related audio quality.
2. Description of the Related Art
In some cases, an analog audio signal is digitized at a prescribed sampling frequency and a prescribed quantization bit number. During the digitizing, signal components having frequencies equal to or higher than a half of the sampling frequency are cut off, and quantization errors occur depending on the quantization bit number. The cutoff of the high-frequency signal components, and the quantization errors cause a reduction in audio quality related to the resultant digital audio signal.
Japanese patent number 3659489 discloses a method of processing a digital audio signal to improve a related audio quality. The digital audio signal has a sequence of samples. In the method of Japanese patent 3659489, a periodically-updated current sample of the digital audio signal and an immediately-preceding sample thereof are compared to detect every local maximum value and every local minimum value of the waveform represented by the digital audio signal. Detection is made as to the number of samples between the position of the occurrence of every local maximum value and the position of the occurrence of a subsequent local minimum value, and the number of samples between the position of the occurrence of every local minimum value and the position of the occurrence of a subsequent local maximum value. Calculation is made as to the difference between every local maximum value and a value represented by a sample immediately-preceding the sample corresponding to the local maximum value, and the difference between every local minimum value and a value represented by a sample immediately-preceding the sample corresponding to the local minimum value. Coefficients are determined in accordance with the detected sample numbers by referring to a coefficient table. The calculated differences and the determined coefficients are multiplied to get a first multiplication result for every local maximum value and a second multiplication result for every local minimum value. The first multiplication result is added to every local maximum value. The second multiplication result is subtracted from the every local minimum value. As a result of the addition and the subtraction, the digital audio signal is corrected.
Japanese patent application publication number 2004-21224 corresponding to US patent application publication number US-2003/0236584 A1 discloses a method of processing a digital audio signal to improve a related audio quality. The digital audio signal has a sequence of samples. In the method of Japanese application 2004-21224, extreme values in an audio waveform represented by the digital audio signal are detected. The extreme values include maximum values and minimum values. An audio frequency represented by the digital audio signal is detected in response to the number of samples between two temporally-adjacent extreme-corresponding samples. A difference between each extreme value and a value of a sample which immediately precedes the present extreme-corresponding sample is calculated. The calculated differences are multiplied by selected one of predetermined coefficients to get corrective values respectively. A decision is made as to whether or not the detected audio frequency is in one selected from predetermined frequency bands. When the detected audio frequency is in the selected frequency band, a corresponding corrective value is added to the maximum value and a corresponding corrective value is subtracted from the minimum value. As a result of the addition and the subtraction, the digital audio signal is corrected.
Usual digital audio signals transmitted in digital television broadcasting or through the Internet result from encoding inclusive of lossy compression that cuts off high-frequency signal components. Examples of such compressively encoding are AAC (Advanced Audio Coding) and MP3. In AAC used by digital television broadcasting, signal components having frequencies of 16 kHz or higher are cut off. On the other hand, MP3 used by portable terminals cuts off signal components having frequencies of 8 kHz or higher. In the case where the method in Japanese patent 3659489 or Japanese application 2004-21224 is applied to the processing of such a compressed digital audio signal (for example, an AAC signal or an MP3 signal), the cut-off of high-frequency signal components can not be sufficiently compensated for. Therefore, a correction-resultant digital audio signal generated by the method tend to appreciably differ from an original analog or digital audio signal in audio quality.
There is a recording medium which stores an analog audio signal recorded many years ago. Generally, the analog audio signal currently reproduced from the recording medium is considerably lower in audio quality than the analog audio signal occurring at the time of the recording. In the case where the method in Japanese patent 3659489 or Japanese application 2004-21224 is applied to the processing of a PCM signal generated by conversion of the currently-reproduced analog audio signal, since only signal samples corresponding to respective extreme values (maximum values and minimum values) are corrected, a correction-resultant digital audio signal appreciably differs in audio quality from the analog audio signal occurring at the time of the recording.
There is an analog audio signal resulting from repetitively dubbing. Generally, the present analog audio signal is considerably lower in audio quality than the original analog audio signal. In the case where the method in Japanese patent 3659489 or Japanese application 2004-21224 is applied to the processing of a PCM signal generated by conversion of the present analog audio signal, since only signal samples corresponding to respective extreme values (maximum values and minimum values) are corrected, a correction-resultant digital audio signal appreciably differs in audio quality from the original analog audio signal.